Networks and origination
| Term | What it means |
|---|---|
| PSTN | Public Switched Telephone Network — the global circuit-switched / IP voice network that terminates real phone numbers. |
| VoIP | Voice over IP — voice carried as packets over data networks instead of circuits. |
| SIP | Session Initiation Protocol (RFC 3261) — the signaling layer that sets up, modifies, and tears down voice sessions. |
| RTP | Real-time Transport Protocol — the media layer that actually carries the audio packets after SIP has set up the call. |
| SRTP | Secure RTP — RTP encrypted with DTLS-SRTP keys negotiated during signaling. WebRTC mandates it. |
| DTLS-SRTP | Datagram TLS used to negotiate SRTP keys. The standard WebRTC media-encryption handshake. |
| ICE / STUN / TURN | NAT-traversal toolkit for peer-to-peer media. ICE picks the candidate path; STUN discovers public addresses; TURN relays media when peer-to-peer is blocked. WebRTC uses all three. |
| SDP | Session Description Protocol — the offer/answer body inside SIP/WebRTC that says “I speak Opus and PCMU on UDP/16384”. |
| DID | Direct Inward Dial — a phone number that routes inbound calls into a specific extension or trunk. “Buy a DID” = rent a phone number. |
| DDI | European synonym for DID. |
| Toll-free number (TFN) | 800/888/etc numbers; the callee pays the per-minute cost. |
| E.164 | The international number format: +<country><number>, max 15 digits. Always quote and store numbers in E.164. |
| CNAM | Caller-ID Name. The string the receiving phone displays. Distinct from the calling number. |
| CLI / Caller-ID | The number presented to the called party. Subject to regulatory rules (US STIR/SHAKEN, EU caller-ID spoofing laws). |
| ANI | Automatic Number Identification — the billing number on the receiving side, used by 911 and toll-free services. |
Trunks and carriers
| Term | What it means |
|---|---|
| SIP trunk | A logical connection between a voice platform and a carrier (or PBX). Has channels, codecs, auth credentials, and an IP allowlist. |
| CPS | Calls Per Second — pacing limit a trunk accepts before throttling. Carriers price by CPS. |
| Channels | Concurrent in-flight calls a trunk can hold. CPS × ACD ≈ steady-state channel usage. |
| SBC | Session Border Controller — a hardened SIP/RTP proxy at the edge of a network. Handles NAT, security, transcoding. |
| PBX | Private Branch eXchange — an enterprise call switch. Sits between phones and trunks. |
| Origination | Outbound calls — your platform → PSTN. |
| Termination | The carrier’s term for “delivering” your outbound call to its destination. Confusingly, also the call’s hangup. |
| Number portability (LNR / LNP) | Ability to keep a phone number when switching carriers. Adds a routing lookup before the call leaves your network. |
| DTMF | The 0–9, *, # tones. Three signaling modes: in-band (audible), RFC 2833 (RTP events), or SIP INFO. RFC 2833 is preferred wherever possible. |
Codecs and media
| Term | What it means |
|---|---|
| G.711 µ-law (PCMU) | 8 kHz, 64 kbit/s codec. North-American PSTN standard. Lossy log compression of 14-bit linear samples. |
| G.711 A-law (PCMA) | Same as µ-law, different log curve. EU/ROW PSTN standard. |
| Opus | Modern wideband codec, 6 kbit/s – 510 kbit/s. WebRTC default. Sounds dramatically better than G.711. |
| G.722 | 16 kHz wideband codec at 64 kbit/s. “HD voice” on a lot of mobile networks. |
| AMR / AMR-WB | Mobile-network codecs. Rare on PSTN handoff. |
| Linear16 / PCM16 | Uncompressed 16-bit PCM. What ML/AI services usually want. |
| Transcoding | Converting one codec to another mid-call. Costs CPU; introduces ≥ 20 ms of latency. |
| Jitter | Variation in packet inter-arrival time. Measured in ms. Anything > 30 ms typically requires de-jitter buffering. |
| Jitter buffer | An adaptive queue that smooths jitter at the cost of latency. Bigger buffer = better audio, more delay. |
| Packet loss | % of RTP packets that never arrive. > 1 % is audible; > 3 % is unintelligible. |
| PLC | Packet Loss Concealment — synthesizes plausible audio to mask a missing packet. Most modern codecs include it. |
| MOS | Mean Opinion Score, 1.0–5.0. Subjective quality. ≥ 4.0 = “toll quality”. |
| R-factor | Objective ITU E-Model score 0–100. Drives an estimated MOS. |
| VAD | Voice Activity Detection. Suppresses silence frames or triggers barge-in. |
| AEC / AGC / NS | Acoustic Echo Cancellation / Auto Gain Control / Noise Suppression — the three pillars of the WebRTC audio-processing pipeline. |
| Comfort noise | Synthetic background hiss inserted during silence to keep the line from sounding “dead”. |
Transports — how the audio actually travels
The four transports any modern voice platform has to know:| Transport | What it is | Where it’s used |
|---|---|---|
| SIP + RTP | Classic VoIP. Signaling on SIP (TCP/UDP/TLS), media on RTP/UDP. The carrier-facing protocol. | Carrier ↔ gateway trunks. |
| WebRTC | Browser-native voice/video. SDP over WebSocket signaling, DTLS-SRTP media over UDP, with ICE for NAT traversal. Mandatory secure, peer-to-peer-by-default media. | Browser softphones, click-to-dial, agent dashboards, embedded contact-centre widgets, integrations with LiveKit / Daily / Chime / Twilio Media Streams / Vapi. |
| WebTransport | HTTP/3 alternative for browser ↔ server bidirectional streams. Same underlying QUIC, no peer-to-peer, no ICE. | Browser apps that want a low-overhead control plane multiplexed with media on the same QUIC connection. |
| QUIC | UDP-based encrypted multiplexed transport. Native SDKs use it directly to talk to the gateway. | Server-side Python / Go / Rust / Java / .NET workloads. |
- SIP/RTP for trunk-side (carrier ↔ gateway). The PSTN doesn’t speak WebRTC.
- WebRTC for last-mile to humans in browsers or vendor SDKs (LiveKit, Daily, Chime, Vapi, Twilio Media Streams).
- QUIC / WebTransport for server-side or browser control workloads where the audio source is already-decoded PCM (e.g. a TTS pipeline) and you want minimum framing overhead.
Call lifecycle and patterns
| Term | What it means |
|---|---|
| A-leg / B-leg | A-leg = caller side; B-leg = callee side. A bridge connects them. |
| Ringback tone | The “ring ring” the caller hears before answer. Generated locally OR by the far end (early media). |
| Early media | Audio sent before the SIP 200 OK. Used for “the number you have dialed…” intercepts. |
| Bridge | Connect two legs so audio flows both ways. |
| Park | Hold a call without bridging — keep the line open with music or silence. |
| Music on hold (MOH) | Audio played to a parked call. Configurable via URL or stream. |
| Transfer (blind / attended) | Industry terms for handing a call to another destination. Not implemented — out of scope on this platform. |
| Conference | Industry term for mixing three or more legs into one audio stream. Not implemented — out of scope on this platform. |
| Whisper / Coach | Industry terms for one-way supervisor audio. Not implemented — the platform’s Barge is unrelated; see the next row. |
| Barge (AI barge-in) | The headline AI-network unblocker. While an AI agent is speaking, the gateway listens for the human caller’s voice on the inbound leg; the moment it crosses the configured energy / VAD / semantic threshold, the AI’s outbound audio is gated and the human is heard immediately. Without it, AI voicebots feel deaf — the human has to wait for the AI to finish before being acknowledged. Configured per-call via auto_barge_in, barge_in_patience_ms, and per-trunk auto_bargein_mode (energy / vad / semantic / off). |
| Barge-in patience | Grace window in milliseconds between detecting the human’s voice and gating the AI. Too short = AI stops on a cough; too long = AI talks over the caller. Typical: 150–350 ms. |
| IVR | Interactive Voice Response — menus driven by DTMF or speech. |
| ACD | Automatic Call Distribution — routing rules that send inbound calls to queues, agents, skill groups. |
| AMD | Answering-Machine Detection — heuristic classifier that says “human” vs “voicemail” within the first ~2 s of audio. |
Routing and dialplan
| Term | What it means |
|---|---|
| Dialplan | The rules engine that decides what happens to a call: hang up, park, play, bridge, hand to AI. |
| LCR | Least-Cost Routing — pick the cheapest carrier per destination prefix. |
| Failover routing | Try carrier A; on 503 Service Unavailable, try carrier B. Trip a circuit breaker after N consecutive failures. |
| Geo-routing | Choose a regional gateway based on caller location, latency, or compliance. |
| Whitelist / Blocklist | Static allow/deny lists per trunk. Often required by carriers for outbound caller-ID. |
| Inbound rule | What to do when a PSTN call arrives at one of your DIDs: REJECT, PLAY-AND-HANGUP, NOTIFY-AND-HANGUP, or HANDLE-AI. |
Compliance vocabulary
| Term | What it means |
|---|---|
| STIR/SHAKEN | US/CA framework for caller-ID attestation. Every outbound INVITE carries an Identity JWT signed by your carrier or platform. Attestation A/B/C controls how trustworthy the receiving carrier marks your call. |
| TCPA | US Telephone Consumer Protection Act. Regulates outbound dialing — quiet hours, abandoned-call rate caps, consent requirements. |
| Do-Not-Call (DNC) | Federal + state lists. DNC suppression must run before every dial. |
| GDPR / CCPA | Privacy regimes that govern call recordings and CDR storage. Affects retention windows. |
| HIPAA | US health-info privacy. Healthcare deployments require a BAA + encrypted recording storage. |
| PCI-DSS | Card-data privacy. If your IVR collects card numbers, the IVR step must redact DTMF from recordings (“DTMF masking”). |
| E911 / 112 | Emergency calling with location attached as PIDF-LO. Mandatory for any platform that lets users dial out. |
| CALEA / Lawful Intercept | Government wiretap obligations for carriers. Honoured at the upstream carrier, not at the gateway. |
Where this maps to TeleQuick
For each section above, the corresponding handle on this platform:| Concept area | Lives in |
|---|---|
| Trunk lifecycle, codec preferences, inbound rules | Admin → Trunks |
| Outbound origination, bulk dialer, hangup, barge | Public RPC — see Method IDs |
| Dialplan actions (park, MOH, playback, transfer) | ExecuteDialplan and bucket actions |
| WebRTC vendor adapters (Browser, LiveKit, Daily, Chime, Twilio, Vapi) | Platform → WebRTC Vendors |
| Compliance headers (STIR/SHAKEN, PANI, PIDF-LO) | Platform → Regulatory |
| MOS, jitter, packet loss, CDRs | Platform → Telemetry |
The four numbers that bound a voice deployment
Any sizing or capacity discussion comes back to these:- CPS sustained (per trunk and aggregate).
- Concurrent channel ceiling.
- End-to-end latency (caller mouth → callee ear, in ms).
- MOS under typical packet-loss conditions.